From 9e6966646b6bc5078d579151b90016522d4ff2cb Mon Sep 17 00:00:00 2001 From: Olek Poplavsky Date: Thu, 24 Jan 2019 23:30:03 -0500 Subject: [PATCH 1/3] ALSA: usb-audio: Add Opus #3 to quirks for native DSD support This patch adds quirk VID/PID IDs for the Opus #3 DAP (made by 'The Bit') in order to enable Native DSD support. [ NOTE: this could be handled in the generic way with fp->dvd_raw if we add 0x10cb to the vendor whitelist, but since 0x10cb shows a different vendor name (Erantech), put to the individual entry at this time -- tiwai ] Signed-off-by: Olek Poplavsky Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ebbadb3a7094e..bb8372833fc22 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1492,6 +1492,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ case USB_ID(0x152a, 0x85de): /* SMSL D1 DAC */ case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ From e190161f96b88ffae870405fd6c3fdd1d2e7f98d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 25 Jan 2019 17:11:32 +0100 Subject: [PATCH 2/3] ALSA: pcm: Fix tight loop of OSS capture stream When the trigger=off is passed for a PCM OSS stream, it sets the start_threshold of the given substream to the boundary size, so that it won't be automatically started. This can be problematic for a capture stream, unfortunately, as detected by syzkaller. The scenario is like the following: - In __snd_pcm_lib_xfer() that is invoked from snd_pcm_oss_read() loop, we have a check whether the stream was already started or the stream can be auto-started. - The function at this check returns 0 with trigger=off since we explicitly disable the auto-start. - The loop continues and repeats calling __snd_pcm_lib_xfer() tightly, which may lead to an RCU stall. This patch fixes the bug by simply allowing the wait for non-started stream in the case of OSS capture. For native usages, it's supposed to be done by the caller side (which is user-space), hence it returns zero like before. (In theory, __snd_pcm_lib_xfer() could wait even for the native API usage cases, too; but I'd like to stay in a safer side for not breaking the existing stuff for now.) Reported-by: syzbot+fbe0496f92a0ce7b786c@syzkaller.appspotmail.com Cc: Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 40013b26f6719..6c99fa8ac5fa1 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2112,6 +2112,13 @@ int pcm_lib_apply_appl_ptr(struct snd_pcm_substream *substream, return 0; } +/* allow waiting for a capture stream that hasn't been started */ +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +#define wait_capture_start(substream) ((substream)->oss.oss) +#else +#define wait_capture_start(substream) false +#endif + /* the common loop for read/write data */ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, void *data, bool interleaved, @@ -2182,7 +2189,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, err = snd_pcm_start(substream); if (err < 0) goto _end_unlock; - } else { + } else if (!wait_capture_start(substream)) { /* nothing to do */ err = 0; goto _end_unlock; From 693abe11aa6b27aed6eb8222162f8fb986325cef Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 29 Jan 2019 15:38:21 +0800 Subject: [PATCH 3/3] ALSA: hda/realtek - Fixed hp_pin no value Fix hp_pin always no value. [More notes on the changes: The hp_pin value that is referred in alc294_hp_init() is always zero at the moment the function gets called, hence this is actually useless as in the current code. And, this kind of init sequence should be called from the codec init callback, instead of the parser function. So, the first fix in this patch to move the call call into its own init_hook. OTOH, this function is needed to be called only once after the boot, and it'd take too long for invoking at each resume (where the init callback gets called). So we add a new flag and invoke this only once as an additional fix. The one case is still not covered, though: S4 resume. But this change itself won't lead to any regression in that regard, so we leave S4 issue as is for now and fix it later. -- tiwai ] Fixes: bde1a7459623 ("ALSA: hda/realtek - Fixed headphone issue for ALC700") Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 78 ++++++++++++++++++++--------------- 1 file changed, 45 insertions(+), 33 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b4f472157ebdf..4139aced63f8a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -117,6 +117,7 @@ struct alc_spec { int codec_variant; /* flag for other variants */ unsigned int has_alc5505_dsp:1; unsigned int no_depop_delay:1; + unsigned int done_hp_init:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -3372,6 +3373,48 @@ static void alc_default_shutup(struct hda_codec *codec) snd_hda_shutup_pins(codec); } +static void alc294_hp_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + int i, val; + + if (!hp_pin) + return; + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + msleep(100); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + alc_update_coef_idx(codec, 0x6f, 0x000f, 0);/* Set HP depop to manual mode */ + alc_update_coefex_idx(codec, 0x58, 0x00, 0x8000, 0x8000); /* HP depop procedure start */ + + /* Wait for depop procedure finish */ + val = alc_read_coefex_idx(codec, 0x58, 0x01); + for (i = 0; i < 20 && val & 0x0080; i++) { + msleep(50); + val = alc_read_coefex_idx(codec, 0x58, 0x01); + } + /* Set HP depop to auto mode */ + alc_update_coef_idx(codec, 0x6f, 0x000f, 0x000b); + msleep(50); +} + +static void alc294_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->done_hp_init) { + alc294_hp_init(codec); + spec->done_hp_init = true; + } + alc_default_init(codec); +} + static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, unsigned int val) { @@ -7373,37 +7416,6 @@ static void alc269_fill_coef(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4, 0, 1<<11); } -static void alc294_hp_init(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; - int i, val; - - if (!hp_pin) - return; - - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - - msleep(100); - - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); - - alc_update_coef_idx(codec, 0x6f, 0x000f, 0);/* Set HP depop to manual mode */ - alc_update_coefex_idx(codec, 0x58, 0x00, 0x8000, 0x8000); /* HP depop procedure start */ - - /* Wait for depop procedure finish */ - val = alc_read_coefex_idx(codec, 0x58, 0x01); - for (i = 0; i < 20 && val & 0x0080; i++) { - msleep(50); - val = alc_read_coefex_idx(codec, 0x58, 0x01); - } - /* Set HP depop to auto mode */ - alc_update_coef_idx(codec, 0x6f, 0x000f, 0x000b); - msleep(50); -} - /* */ static int patch_alc269(struct hda_codec *codec) @@ -7529,7 +7541,7 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC294; spec->gen.mixer_nid = 0; /* ALC2x4 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); /* UAJ MIC Vref control by verb */ - alc294_hp_init(codec); + spec->init_hook = alc294_init; break; case 0x10ec0300: spec->codec_variant = ALC269_TYPE_ALC300; @@ -7541,7 +7553,7 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC700; spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ - alc294_hp_init(codec); + spec->init_hook = alc294_init; break; }